The present invention relates generally to Voice over Internet Protocol (VoIP) and, more particularly, to adjusting the volume for multiple VoIP streams.
In VoIP telephony a speaker speaks into a handset (or other receiving device) and a stream of digital information is created there from. This digital stream is transmitted through a VoIP server to a receiver or a call (typically another person to whom the call has been placed). The receiver's handset (or other reception device connected thereto) decodes the digital stream and produces an analog output that ultimately drives a speaker audible to the receiver. Of course, the VoIP server may perform more than just “pass-through” routing. For instance, the server may convert the digital stream from one format to another if the speaker and the receiver are utilizing different types of handsets or handsets that support different protocols.
When connected to a VoIP (or traditional) conference call where multiple speakers are speaking, different voice types and connection types can cause different speakers to have dramatically different volumes. This may lead to difficulties in hearing some speakers relative to other speakers. For instance, one speaker may speak quieter than another speaker and it may be difficult to hear that speaker speak.
In the VoIP environment, one solution is to raise or lower the reception volume of all incoming audio streams at the receiver. That is, each stream is amplified or reduced by the VoIP handset by the same amount. One problem with this solution is that because the volume of all speakers are adjusted by the same amount, a quiet speaker may still sound quite when compared to a loud speaker.
Another solution includes having participants amplify or reduce the transmit volume of their handsets so that all persons are similarly loud. This produces good results on the receiver side when adequately adjusted, but the configuration amongst many participants can be tedious.